1. Field of the Invention
The present invention relates to call processing and is more particularly related to establishing a voice call over a packet switched network via a voice response unit.
2. Discussion of the Background
The popularity and convenience of the Internet has resulted in the reinvention of traditional telephony services. These services are offered over a packet switched network with minimal or no cost to the users. IP (Internet Protocol) telephony, thus, have found significant success, particularly in the long distance market. In general, IP telephony, which is also referred to as Voice-over-IP (VOIP), is the conversion of voice information into data packets that are transmitted over an IP network. Users also have turned to IP telephony as a matter of convenience in that both voice and data services are accessible through a single piece of equipment, namely a personal computer. The continual integration of voice and data services further fuels this demand for IP telephone applications.
With the growing acceptance of IP telephony among the millions of consumers, service providers are cognizant of the impact that these users have on network capacity (e.g., switch sizing, line capacity) as well as network resources (e.g., peripheral voice processing devices). A valuable network resource is the voice response unit (VRU), which provides announcement and interactive voice response functions. These functions have become essential for the expedient treatment of voice calls, especially in call center applications and operator assistance. Because VRU ports are expensive, it is desirable to ensure efficient use of such ports.
FIG. 8 illustrates a conventional IP telephony system. In this system 800, an end office 801 houses a switch 803 and a VRU 805; the switch 803 communicates with the VRU over a release line trunk (RLT). Switch 803 serves user 807 to a public switch telephone network (PSTN) 809. The VRU 805 is not functionally integrated with the IP network 815. That is, the VRU 805 works primarily in conjunction with the switch 803 within the PSTN realm. Using plain old telephone service (POTS), a calling party 807 can place a telephone call over PSTN 809 to a called party 811 or 813.
The PSTN 809 is connected to an IP (Internet Protocol) network 815, thereby enabling communication among the voice stations 807, 811, and 813, which are connected to the public switch telephone network 809, and the personal computers 817 and 819, which are attached to the IP network 815. Attention is now drawn to transmission of voice calls over the IP network 815.
Four possible scenarios exist with the placement of a VOIP call: (1) phone-to-phone, (2) phone-to-PC, (3) PC-to-phone, and (4) PC-to-PC. In the first scenario of phone-to-phone call establishment, voice station 807 is switched through PSTN 809 by switch 803 to a VOIP gateway (not shown), which forwards the call through the IP network 815. The packetized voice call is then routed through the IP network 815, exiting the IP network 815 at an appropriate point to enter PSTN 809 and terminates at voice station 811. Under the second scenario, voice station 807 places a call to personal computer (PC) 817 through switch 803 to PSTN 809. This voice call is then switched by the PSTN 809 to a VOIP gateway (not shown), which forwards the voice call to PC 817 via IP network 815. The third scenario involves PC 817 placing a call to voice station 813, for example. Using a voice encoder, PC 817 introduces a stream of voice packets into IP network 815 that are destined for a VOIP gateway (not shown). A VOIP gateway (not shown) converts the packetized voice information into a POTS electrical signal, which is circuit switched to voice station 813. Lastly, in the fourth scenario, PC 817 establishes a voice call with PC 819. In this case, packetized voice data is transmitted from PC 817 via IP network 815 to PC 819, where the packetized voice data is decoded.
As indicated above, a network resource that permits the efficient processing of voice calls is a VRU 805. FIG. 9 shows a conventional call path that is established by switch 803 to VRU 805. RLT links 901 connect switch 803 to VRU 805, consuming two ports of each of these network components 803 and 805. RLT links 901 enable the release of a call back to switch 803 from VRU 805. This releasing functionality allows the VRU 805 to be dropped from the voice call without impacting the call completion between call originator 903 and call terminator 905.
For explanatory purposes, it is assumed that a VRU 805 is needed to assist with call processing from call originator 903 (i.e., calling party) to call terminator 805 (i.e.. called party). Call originator 903 places a voice call to switch 803 using port 1. In turn, the switch 803 switches the call out of port 3 to port 1 of VRU 805. Once this call is established with VRU 805, the VRU 805 prompts the call originator 903, for example, to collect digits regarding account codes or billing information in order to authorize and validate the call originator 903. After this process, the VRU 805 loops the voice call back to the switch 803 via port 2 over RLT 901 into port 4 of switch 803. Switch 803 then switches the call out of port 2 to call terminator 905. The RLT links 901 permits the VRU 805 to drop out of the call when the call is completed between call originator 903 and call terminator 905. This release mechanism occurs over the PSTN 809. Such a mechanism is important because it frees up the VRU 805 to process other calls; in addition, the switch 803 frees up two of its ports. An equivalent functionality is desirable in an IP telephony system.
Based on the foregoing, there is a clear need for improved approaches for call processing with respect to use of network resources.
There is also a need to increase the integration of voice services over a data network.
There is a further need to minimize the cost of network operation.
Based on the need to efficiently employ network resources, an approach for optimizing the use of VRU in an IP telephony environment is highly desirable.